Hearing aids are portable hearing devices which are used to supply the hard-of-hearing. To accommodate the numerous individual needs, different hearing aid constructions, such as behind-the-ear (BTE) hearing aids, in-the-ear hearing aids (ITE) and concha hearing aids are provided. The hearing aids listed by way of example are worn on the outer ear or in the auditory canal. There are however also bone conduction hearing aids, implantable or vibrotactile hearing aids available on the market. In this case, the damaged ear is stimulated either mechanically or electrically.
Basically hearing aids possess as fundamental components an input transducer, an amplifier and an output transducer. The input transducer is usually a receiving transducer, for example a microphone, and/or an electromagnetic receiver, for example an induction coil. The output transducer is usually implemented as an electroacoustic transducer, for example a miniature loudspeaker, or as an electromechnical transducer, for example bone conduction earpiece. The amplifier is conventionally integrated in a signal processing unit. FIG. 1 shows this basic construction using the example of a behind-the-ear hearing aid. One or more microphone(s) 2 for receiving the sound from the environment are fitted into a hearing aid casing 1 for wearing being the ear. A signal processing unit 3, which is also integrated in the hearing aid casing 1, processes the microphone signals and amplifies them. The output signal of the signal processing unit 3 is transmitted to a loudspeaker or earpiece 4 which outputs an acoustic signal. The sound is optionally transmitted via an acoustic tube, which is fixed by an otoplastic in the auditory canal, to the ear drum of the hearing aid wearer. The power supply to the hearing aid, and in particular that of the signal processing unit 3, is provided by a battery 5 that is also integrated in the hearing aid casing 1.
Audio signals have a specific dynamic range which characterizes the difference between the lowest and the highest levels. Naturally occurring audio signals usually have a high dynamic range while audio equipment, such as radio equipment and hearing aids, have a much lower dynamic range in their output signal. For this reason a dynamic range compression is carried out in the case of said devices or corresponding processing methods.
Dynamic range compression is a non-linear method. Audible distortions therefore occur, in particular in the case of fast recovery time constants, which lead to a reduction in sound quality. The “effective compression” also decreases with increasing modulation frequency owing to the inertia of the control system. For natural signals it is therefore almost impossible to predict the effective compression rate or purposefully adjust it as a function of the modulation frequency. The first problem of distortions in the case of fast recovery time constants can be avoided by appropriately slow control. The second problem of inertia in the control system if anything requires fast control however, and this contradicts said solution to the first problem.
A coherent demodulation for obtaining a complex envelope is known from the article “PROPERTIES FOR MODULATION SPECTRAL FILTERING”, Qin Li and Les Atlas, ICASSP 2005, pages 521 to 524.
From patent specification DE 197 03 228 B4 a method for amplifying input signals of a hearing aid is known in which to ensure dynamic range compression, in addition to detecting the signal level of the input signal, a modulation frequency analysis is carried out.
Document DE 10 2004 044 565 A1 also describes a method for limiting the dynamic range of audio signals. In this case the dynamic range limitation is regulated as a function of the instantaneous frequency of the audio signal for compression.